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Summary:ASTERISK-12341: Call blocked after 1 minute 45 seconde
Reporter:olivier1010 (olivier1010)Labels:
Date Opened:2008-07-08 09:28:36Date Closed:2011-06-07 14:00:47
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:When doing calls between two extensions (G711a codec), the call is dropped after 1m45sec on the called phone.

Then the calling phone hangup the call at about 2m05sec.

After the called phone hangup, issuing a "sip show channels" gives the two channels actives.

192.168.200.60   41          2bb589e920880fe  0x1000 (g722)    No       Rx: ACK                  
192.168.200.40   40          a038a43dc9b9160  0x1000 (g722)    No       Rx: ACK  

After the second phone hangup, the channels are effectively destroyed.

This is perfectly reproductible.

Using Aastra 57i phones firmware 2.2.1. and FreePBX 2.4.1.


i've notified this registration error, this is strange because the phone does register correctly at beginning :

<--- SIP read from UDP://192.168.200.60:5060 --->
REGISTER sip:192.168.200.240 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK0eebc1279e9885cab
Max-Forwards: 70
From: <sip:43@192.168.200.240>;tag=b9a1ec5bcb
To: <sip:43@192.168.200.240>
Call-ID: 099bee27fef189dc
CSeq: 20235 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username="43",realm="asterisk",nonce="33b166a6",uri="sip:192.168.200.240",response="eba83d472e4b97fc34e7bf6272926704",algorithm=MD5
Contact: <sip:43@192.168.200.60:5060;transport=udp>
User-Agent: Aastra 57i/2.2.1.25
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.200.60 : 5060 (NAT)

<--- Transmitting (no NAT) to 192.168.200.60:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK0eebc1279e9885cab;received=192.168.200.60
From: <sip:43@192.168.200.240>;tag=b9a1ec5bcb
To: <sip:43@192.168.200.240>;tag=as175efbde
Call-ID: 099bee27fef189dc
CSeq: 20235 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7354d4ea"
Content-Length: 0




Comments:By: Joshua C. Colp (jcolp) 2008-07-10 18:10:52

Please provide complete console output and a sip debug.
By: Tilghman Lesher (tilghman) 2008-09-11 18:12:29

No response from reporter